Audio playback systems utilise digital data for the playback of (audio) sound recordings. However, audio output equipment, such as speakers, headphones and the like require an analog audio signal in order to function. Accordingly, in an audio playback system, a digital-to-analogue converter (DAC) is required to convert digital data into an analog audio signal.
Digital data corresponding to an analog audio signal can be provided to a DAC in a number of different formats. For example, one known format of digital data is Compact Disc Digital Audio. This format is a two-channel, 16-bit signal with pulse code modulation (PCM), and a 44.1 kHz sampling rate per channel. Another known data format is Direct Stream Digital™ (DSD). DSD is a 1-bit bitstream data format which uses pulse-density modulation.
Whatever choice of data format is provided as an input to the DAC, the role of the DAC is to convert a data signal into an analog audio signal as accurately as possible. This accuracy may be expressed in terms of a signal-to-noise ratio (SNR), with inaccuracy being equivalent to the introduction of a noise component by the DAC.
One main source of noise in digital signals is Intersymbol Interference (ISI). ISI is a form of distortion which occurs when one symbol interferes with subsequent symbols in a digital signal.
U.S. Pat. No. 6,061,010 discloses that in a DAC, ISI occurs when the output waveform for a particular clock period is a function not only of the digital code applied to the DAC for that clock period, but also of the digital code applied for a preceding clock period. This interference can cause both distortion and noise to appear at the DAC output.
U.S. Pat. No. 6,061,010 further discloses that in a DAC, ISI manifests itself by the area under the DAC output waveform for a given clock period (i.e. the time integral of the output), depending partly on the applied digital code (i.e. input signal value) during the previous clock period. In such a DAC, the accuracy of the area under the output waveform curve for a selected output sample is an extremely important measure of performance, as it contributes heavily to the purity of the low-frequency part of the output spectrum. That is, the output of an n-bit DAC comprises the sum of “n” analog waveforms, taking the form of a voltage or a current. In order for the DAC output to be free of ISI, each individual constituent analog bit waveform thus must be free of ISI (the output being a linear summation of the constituent waveforms).
In particular U.S. Pat. No. 6,061,010 discloses that ISI may result from unequal rise and fall times in the most significant bit output current. Conventionally, this kind of ISI is reduced by forcing the output bit to start from zero, reach its final value, and return to zero all within a single bit clock period (clock cycle). This is called a “return-to zero” (RTZ) code. Since there is a rise and a fall within every clock cycle, the area under each waveform pulse is guaranteed to be independent from prior bit values.
However, U.S. Pat. No. 6,061,010 explains that the RTZ approach is not without its limitations. The RTZ approach introduces full scale steps into the output waveform. This potentially degrades performance or causes problems in two ways. First, the operation of a circuit connected to receive the DAC output may become non-linear when driven by such large, high-speed steps. Second, any error in the clock edge timing, due to jitter or other mechanisms, may produce a large error in the output due to the large step size. In oversampled DACs, this is a particularly egregious problem, because sample-to-sample output current or voltage differences normally would be a small fraction of the full-scale range of the converter; however using an RTZ scheme, the average step size may be dramatically larger and the sensitivity to clock jitter may therefore be degraded seriously.
Thus the present inventor(s) propose the present invention with the knowledge that there is a need for a converter for converting bitstream (digital) signals to audio signals which has a high SNR.